f20b67173c
This commit abandons the way the specifications state to quantize the coefficients, makes use of the new LPC float functions and is much better. The original way of converting non-normalized float samples to int32_t which out LPC system expects was wrong and it was wrong to assume the coefficients that are generated are also valid. It was essentially a full garbage-in, garbage-out system and it definitely shows when looking at spectrals and listening. The high frequencies were very overattenuated. The new LPC function performs the analysis directly. The specifications state to quantize the coefficients into four bit index values using an asin() function which of course had to have ugly ternary operators because the function turns negative if the coefficients are negative which when encoding causes invalid bitstream to get generated. This deviates from this by using the direct TNS tables, which are fairly small since you only have 4 bits at most for index values. The LPC values are directly quantized against the tables and are then used to perform filtering after the requantization, which simply fetches the array values. The end result is that TNS works much better now and doesn't attenuate anything but the actual signal, e.g. TNS removes quantization errors and does it's job correctly now. It might be enabled by default soon since it doesn't hurt and helps reduce nastyness at low bitrates. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
893 lines
34 KiB
C
893 lines
34 KiB
C
/*
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* AAC encoder
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* Copyright (C) 2008 Konstantin Shishkov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* AAC encoder
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*/
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/***********************************
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* TODOs:
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* add sane pulse detection
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* add temporal noise shaping
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***********************************/
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#include "libavutil/float_dsp.h"
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "put_bits.h"
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#include "internal.h"
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#include "mpeg4audio.h"
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#include "kbdwin.h"
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#include "sinewin.h"
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#include "aac.h"
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#include "aactab.h"
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#include "aacenc.h"
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#include "aacenctab.h"
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#include "aacenc_utils.h"
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#include "psymodel.h"
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/**
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* Make AAC audio config object.
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* @see 1.6.2.1 "Syntax - AudioSpecificConfig"
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*/
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static void put_audio_specific_config(AVCodecContext *avctx)
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{
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PutBitContext pb;
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AACEncContext *s = avctx->priv_data;
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init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
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put_bits(&pb, 5, s->profile+1); //profile
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put_bits(&pb, 4, s->samplerate_index); //sample rate index
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put_bits(&pb, 4, s->channels);
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//GASpecificConfig
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put_bits(&pb, 1, 0); //frame length - 1024 samples
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put_bits(&pb, 1, 0); //does not depend on core coder
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put_bits(&pb, 1, 0); //is not extension
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//Explicitly Mark SBR absent
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put_bits(&pb, 11, 0x2b7); //sync extension
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put_bits(&pb, 5, AOT_SBR);
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put_bits(&pb, 1, 0);
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flush_put_bits(&pb);
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}
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#define WINDOW_FUNC(type) \
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static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
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SingleChannelElement *sce, \
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const float *audio)
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WINDOW_FUNC(only_long)
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{
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const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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float *out = sce->ret_buf;
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fdsp->vector_fmul (out, audio, lwindow, 1024);
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fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
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}
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WINDOW_FUNC(long_start)
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{
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const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
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float *out = sce->ret_buf;
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fdsp->vector_fmul(out, audio, lwindow, 1024);
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memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
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fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
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memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
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}
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WINDOW_FUNC(long_stop)
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{
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const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
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float *out = sce->ret_buf;
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memset(out, 0, sizeof(out[0]) * 448);
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fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
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memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
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fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
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}
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WINDOW_FUNC(eight_short)
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{
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const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
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const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
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const float *in = audio + 448;
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float *out = sce->ret_buf;
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int w;
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for (w = 0; w < 8; w++) {
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fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
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out += 128;
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in += 128;
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fdsp->vector_fmul_reverse(out, in, swindow, 128);
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out += 128;
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}
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}
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static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
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SingleChannelElement *sce,
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const float *audio) = {
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[ONLY_LONG_SEQUENCE] = apply_only_long_window,
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[LONG_START_SEQUENCE] = apply_long_start_window,
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[EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
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[LONG_STOP_SEQUENCE] = apply_long_stop_window
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};
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static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
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float *audio)
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{
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int i;
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float *output = sce->ret_buf;
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apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
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if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
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s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
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else
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for (i = 0; i < 1024; i += 128)
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s->mdct128.mdct_calc(&s->mdct128, &sce->coeffs[i], output + i*2);
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memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
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memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
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}
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/**
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* Encode ics_info element.
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* @see Table 4.6 (syntax of ics_info)
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*/
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static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
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{
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int w;
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put_bits(&s->pb, 1, 0); // ics_reserved bit
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put_bits(&s->pb, 2, info->window_sequence[0]);
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put_bits(&s->pb, 1, info->use_kb_window[0]);
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if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
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put_bits(&s->pb, 6, info->max_sfb);
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put_bits(&s->pb, 1, !!info->predictor_present);
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} else {
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put_bits(&s->pb, 4, info->max_sfb);
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for (w = 1; w < 8; w++)
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put_bits(&s->pb, 1, !info->group_len[w]);
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}
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}
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/**
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* Encode MS data.
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* @see 4.6.8.1 "Joint Coding - M/S Stereo"
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*/
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static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
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{
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int i, w;
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put_bits(pb, 2, cpe->ms_mode);
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if (cpe->ms_mode == 1)
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for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
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for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
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put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
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}
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/**
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* Produce integer coefficients from scalefactors provided by the model.
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*/
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static void adjust_frame_information(ChannelElement *cpe, int chans)
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{
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int i, w, w2, g, ch;
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int maxsfb, cmaxsfb;
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IndividualChannelStream *ics;
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if (cpe->common_window) {
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ics = &cpe->ch[0].ics;
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for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
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for (w2 = 0; w2 < ics->group_len[w]; w2++) {
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int start = (w+w2) * 128;
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for (g = 0; g < ics->num_swb; g++) {
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//apply Intensity stereo coeffs transformation
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if (cpe->is_mask[w*16 + g]) {
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int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
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float scale = cpe->ch[0].is_ener[w*16+g];
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for (i = 0; i < ics->swb_sizes[g]; i++) {
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cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i]) * scale;
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cpe->ch[1].coeffs[start+i] = 0.0f;
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}
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} else if (cpe->ms_mask[w*16 + g] &&
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cpe->ch[0].band_type[w*16 + g] < NOISE_BT &&
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cpe->ch[1].band_type[w*16 + g] < NOISE_BT) {
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for (i = 0; i < ics->swb_sizes[g]; i++) {
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float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
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float R = L - cpe->ch[1].coeffs[start+i];
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cpe->ch[0].coeffs[start+i] = L;
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cpe->ch[1].coeffs[start+i] = R;
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}
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}
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start += ics->swb_sizes[g];
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}
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}
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}
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}
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for (ch = 0; ch < chans; ch++) {
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IndividualChannelStream *ics = &cpe->ch[ch].ics;
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maxsfb = 0;
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cpe->ch[ch].pulse.num_pulse = 0;
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for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
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for (w2 = 0; w2 < ics->group_len[w]; w2++) {
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for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
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;
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maxsfb = FFMAX(maxsfb, cmaxsfb);
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}
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}
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ics->max_sfb = maxsfb;
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//adjust zero bands for window groups
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for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
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for (g = 0; g < ics->max_sfb; g++) {
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i = 1;
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for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
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if (!cpe->ch[ch].zeroes[w2*16 + g]) {
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i = 0;
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break;
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}
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}
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cpe->ch[ch].zeroes[w*16 + g] = i;
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}
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}
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}
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if (chans > 1 && cpe->common_window) {
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IndividualChannelStream *ics0 = &cpe->ch[0].ics;
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IndividualChannelStream *ics1 = &cpe->ch[1].ics;
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int msc = 0;
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ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
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ics1->max_sfb = ics0->max_sfb;
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for (w = 0; w < ics0->num_windows*16; w += 16)
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for (i = 0; i < ics0->max_sfb; i++)
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if (cpe->ms_mask[w+i])
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msc++;
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if (msc == 0 || ics0->max_sfb == 0)
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cpe->ms_mode = 0;
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else
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cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
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}
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}
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/**
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* Encode scalefactor band coding type.
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*/
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static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
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{
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int w;
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
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s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
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}
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/**
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* Encode scalefactors.
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*/
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static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
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SingleChannelElement *sce)
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{
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int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
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int off_is = 0, noise_flag = 1;
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int i, w;
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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for (i = 0; i < sce->ics.max_sfb; i++) {
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if (!sce->zeroes[w*16 + i]) {
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if (sce->band_type[w*16 + i] == NOISE_BT) {
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diff = sce->sf_idx[w*16 + i] - off_pns;
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off_pns = sce->sf_idx[w*16 + i];
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if (noise_flag-- > 0) {
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put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
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continue;
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}
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} else if (sce->band_type[w*16 + i] == INTENSITY_BT ||
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sce->band_type[w*16 + i] == INTENSITY_BT2) {
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diff = sce->sf_idx[w*16 + i] - off_is;
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off_is = sce->sf_idx[w*16 + i];
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} else {
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diff = sce->sf_idx[w*16 + i] - off_sf;
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off_sf = sce->sf_idx[w*16 + i];
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}
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diff += SCALE_DIFF_ZERO;
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av_assert0(diff >= 0 && diff <= 120);
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put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
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}
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}
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}
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}
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/**
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* Encode pulse data.
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*/
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static void encode_pulses(AACEncContext *s, Pulse *pulse)
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{
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int i;
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put_bits(&s->pb, 1, !!pulse->num_pulse);
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if (!pulse->num_pulse)
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return;
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put_bits(&s->pb, 2, pulse->num_pulse - 1);
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put_bits(&s->pb, 6, pulse->start);
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for (i = 0; i < pulse->num_pulse; i++) {
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put_bits(&s->pb, 5, pulse->pos[i]);
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put_bits(&s->pb, 4, pulse->amp[i]);
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}
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}
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/**
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* Encode spectral coefficients processed by psychoacoustic model.
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*/
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static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
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{
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int start, i, w, w2;
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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start = 0;
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for (i = 0; i < sce->ics.max_sfb; i++) {
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if (sce->zeroes[w*16 + i]) {
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start += sce->ics.swb_sizes[i];
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continue;
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}
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for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
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s->coder->quantize_and_encode_band(s, &s->pb,
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&sce->coeffs[start + w2*128],
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NULL, sce->ics.swb_sizes[i],
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sce->sf_idx[w*16 + i],
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sce->band_type[w*16 + i],
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s->lambda,
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sce->ics.window_clipping[w]);
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}
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start += sce->ics.swb_sizes[i];
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}
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}
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}
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/**
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* Downscale spectral coefficients for near-clipping windows to avoid artifacts
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*/
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static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
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{
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int start, i, j, w;
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if (sce->ics.clip_avoidance_factor < 1.0f) {
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for (w = 0; w < sce->ics.num_windows; w++) {
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start = 0;
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for (i = 0; i < sce->ics.max_sfb; i++) {
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float *swb_coeffs = &sce->coeffs[start + w*128];
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for (j = 0; j < sce->ics.swb_sizes[i]; j++)
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swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
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start += sce->ics.swb_sizes[i];
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}
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}
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}
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}
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/**
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* Encode one channel of audio data.
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*/
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static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
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SingleChannelElement *sce,
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int common_window)
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{
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put_bits(&s->pb, 8, sce->sf_idx[0]);
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if (!common_window) {
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put_ics_info(s, &sce->ics);
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if (s->coder->encode_main_pred)
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s->coder->encode_main_pred(s, sce);
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}
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encode_band_info(s, sce);
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encode_scale_factors(avctx, s, sce);
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encode_pulses(s, &sce->pulse);
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put_bits(&s->pb, 1, !!sce->tns.present);
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if (s->coder->encode_tns_info)
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s->coder->encode_tns_info(s, sce);
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put_bits(&s->pb, 1, 0); //ssr
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encode_spectral_coeffs(s, sce);
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return 0;
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}
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/**
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* Write some auxiliary information about the created AAC file.
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*/
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static void put_bitstream_info(AACEncContext *s, const char *name)
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{
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int i, namelen, padbits;
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namelen = strlen(name) + 2;
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put_bits(&s->pb, 3, TYPE_FIL);
|
|
put_bits(&s->pb, 4, FFMIN(namelen, 15));
|
|
if (namelen >= 15)
|
|
put_bits(&s->pb, 8, namelen - 14);
|
|
put_bits(&s->pb, 4, 0); //extension type - filler
|
|
padbits = -put_bits_count(&s->pb) & 7;
|
|
avpriv_align_put_bits(&s->pb);
|
|
for (i = 0; i < namelen - 2; i++)
|
|
put_bits(&s->pb, 8, name[i]);
|
|
put_bits(&s->pb, 12 - padbits, 0);
|
|
}
|
|
|
|
/*
|
|
* Copy input samples.
|
|
* Channels are reordered from libavcodec's default order to AAC order.
|
|
*/
|
|
static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
|
|
{
|
|
int ch;
|
|
int end = 2048 + (frame ? frame->nb_samples : 0);
|
|
const uint8_t *channel_map = aac_chan_maps[s->channels - 1];
|
|
|
|
/* copy and remap input samples */
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
/* copy last 1024 samples of previous frame to the start of the current frame */
|
|
memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
|
|
|
|
/* copy new samples and zero any remaining samples */
|
|
if (frame) {
|
|
memcpy(&s->planar_samples[ch][2048],
|
|
frame->extended_data[channel_map[ch]],
|
|
frame->nb_samples * sizeof(s->planar_samples[0][0]));
|
|
}
|
|
memset(&s->planar_samples[ch][end], 0,
|
|
(3072 - end) * sizeof(s->planar_samples[0][0]));
|
|
}
|
|
}
|
|
|
|
static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
|
|
const AVFrame *frame, int *got_packet_ptr)
|
|
{
|
|
AACEncContext *s = avctx->priv_data;
|
|
float **samples = s->planar_samples, *samples2, *la, *overlap;
|
|
ChannelElement *cpe;
|
|
SingleChannelElement *sce;
|
|
int i, ch, w, g, chans, tag, start_ch, ret;
|
|
int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
|
|
int chan_el_counter[4];
|
|
FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
|
|
|
|
if (s->last_frame == 2)
|
|
return 0;
|
|
|
|
/* add current frame to queue */
|
|
if (frame) {
|
|
if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
copy_input_samples(s, frame);
|
|
if (s->psypp)
|
|
ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
|
|
|
|
if (!avctx->frame_number)
|
|
return 0;
|
|
|
|
start_ch = 0;
|
|
for (i = 0; i < s->chan_map[0]; i++) {
|
|
FFPsyWindowInfo* wi = windows + start_ch;
|
|
tag = s->chan_map[i+1];
|
|
chans = tag == TYPE_CPE ? 2 : 1;
|
|
cpe = &s->cpe[i];
|
|
for (ch = 0; ch < chans; ch++) {
|
|
IndividualChannelStream *ics = &cpe->ch[ch].ics;
|
|
int cur_channel = start_ch + ch;
|
|
float clip_avoidance_factor;
|
|
overlap = &samples[cur_channel][0];
|
|
samples2 = overlap + 1024;
|
|
la = samples2 + (448+64);
|
|
if (!frame)
|
|
la = NULL;
|
|
if (tag == TYPE_LFE) {
|
|
wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
|
|
wi[ch].window_shape = 0;
|
|
wi[ch].num_windows = 1;
|
|
wi[ch].grouping[0] = 1;
|
|
|
|
/* Only the lowest 12 coefficients are used in a LFE channel.
|
|
* The expression below results in only the bottom 8 coefficients
|
|
* being used for 11.025kHz to 16kHz sample rates.
|
|
*/
|
|
ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
|
|
} else {
|
|
wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,
|
|
ics->window_sequence[0]);
|
|
}
|
|
ics->window_sequence[1] = ics->window_sequence[0];
|
|
ics->window_sequence[0] = wi[ch].window_type[0];
|
|
ics->use_kb_window[1] = ics->use_kb_window[0];
|
|
ics->use_kb_window[0] = wi[ch].window_shape;
|
|
ics->num_windows = wi[ch].num_windows;
|
|
ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
|
|
ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
|
|
ics->swb_offset = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
|
|
ff_swb_offset_128 [s->samplerate_index]:
|
|
ff_swb_offset_1024[s->samplerate_index];
|
|
clip_avoidance_factor = 0.0f;
|
|
for (w = 0; w < ics->num_windows; w++)
|
|
ics->group_len[w] = wi[ch].grouping[w];
|
|
for (w = 0; w < ics->num_windows; w++) {
|
|
if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
|
|
ics->window_clipping[w] = 1;
|
|
clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
|
|
} else {
|
|
ics->window_clipping[w] = 0;
|
|
}
|
|
}
|
|
if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
|
|
ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
|
|
} else {
|
|
ics->clip_avoidance_factor = 1.0f;
|
|
}
|
|
|
|
apply_window_and_mdct(s, &cpe->ch[ch], overlap);
|
|
if (isnan(cpe->ch->coeffs[0])) {
|
|
av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
avoid_clipping(s, &cpe->ch[ch]);
|
|
}
|
|
start_ch += chans;
|
|
}
|
|
if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels, 0)) < 0)
|
|
return ret;
|
|
do {
|
|
int frame_bits;
|
|
|
|
init_put_bits(&s->pb, avpkt->data, avpkt->size);
|
|
|
|
if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
|
|
put_bitstream_info(s, LIBAVCODEC_IDENT);
|
|
start_ch = 0;
|
|
memset(chan_el_counter, 0, sizeof(chan_el_counter));
|
|
for (i = 0; i < s->chan_map[0]; i++) {
|
|
FFPsyWindowInfo* wi = windows + start_ch;
|
|
const float *coeffs[2];
|
|
tag = s->chan_map[i+1];
|
|
chans = tag == TYPE_CPE ? 2 : 1;
|
|
cpe = &s->cpe[i];
|
|
memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
|
|
memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
|
|
put_bits(&s->pb, 3, tag);
|
|
put_bits(&s->pb, 4, chan_el_counter[tag]++);
|
|
for (ch = 0; ch < chans; ch++) {
|
|
sce = &cpe->ch[ch];
|
|
coeffs[ch] = sce->coeffs;
|
|
sce->ics.predictor_present = 0;
|
|
memset(&sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
|
|
memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
|
|
for (w = 0; w < 128; w++)
|
|
if (sce->band_type[w] > RESERVED_BT)
|
|
sce->band_type[w] = 0;
|
|
}
|
|
s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
|
|
for (ch = 0; ch < chans; ch++) {
|
|
s->cur_channel = start_ch + ch;
|
|
s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
|
|
}
|
|
cpe->common_window = 0;
|
|
if (chans > 1
|
|
&& wi[0].window_type[0] == wi[1].window_type[0]
|
|
&& wi[0].window_shape == wi[1].window_shape) {
|
|
|
|
cpe->common_window = 1;
|
|
for (w = 0; w < wi[0].num_windows; w++) {
|
|
if (wi[0].grouping[w] != wi[1].grouping[w]) {
|
|
cpe->common_window = 0;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
for (ch = 0; ch < chans; ch++) {
|
|
sce = &cpe->ch[ch];
|
|
s->cur_channel = start_ch + ch;
|
|
if (s->options.pns && s->coder->search_for_pns)
|
|
s->coder->search_for_pns(s, avctx, sce);
|
|
if (s->options.tns && s->coder->search_for_tns)
|
|
s->coder->search_for_tns(s, sce);
|
|
if (s->options.tns && s->coder->apply_tns_filt)
|
|
s->coder->apply_tns_filt(sce);
|
|
if (sce->tns.present)
|
|
tns_mode = 1;
|
|
}
|
|
s->cur_channel = start_ch;
|
|
if (s->options.stereo_mode && cpe->common_window) {
|
|
if (s->options.stereo_mode > 0) {
|
|
IndividualChannelStream *ics = &cpe->ch[0].ics;
|
|
for (w = 0; w < ics->num_windows; w += ics->group_len[w])
|
|
for (g = 0; g < ics->num_swb; g++)
|
|
cpe->ms_mask[w*16+g] = 1;
|
|
} else if (s->coder->search_for_ms) {
|
|
s->coder->search_for_ms(s, cpe);
|
|
}
|
|
}
|
|
if (s->options.intensity_stereo && s->coder->search_for_is) {
|
|
s->coder->search_for_is(s, avctx, cpe);
|
|
if (cpe->is_mode) is_mode = 1;
|
|
}
|
|
if (s->coder->set_special_band_scalefactors)
|
|
for (ch = 0; ch < chans; ch++)
|
|
s->coder->set_special_band_scalefactors(s, &cpe->ch[ch]);
|
|
adjust_frame_information(cpe, chans);
|
|
for (ch = 0; ch < chans; ch++) {
|
|
sce = &cpe->ch[ch];
|
|
s->cur_channel = start_ch + ch;
|
|
if (s->options.pred && s->coder->search_for_pred)
|
|
s->coder->search_for_pred(s, sce);
|
|
if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
|
|
}
|
|
if (s->options.pred && s->coder->adjust_common_prediction)
|
|
s->coder->adjust_common_prediction(s, cpe);
|
|
for (ch = 0; ch < chans; ch++) {
|
|
sce = &cpe->ch[ch];
|
|
s->cur_channel = start_ch + ch;
|
|
if (s->options.pred && s->coder->apply_main_pred)
|
|
s->coder->apply_main_pred(s, sce);
|
|
}
|
|
s->cur_channel = start_ch;
|
|
if (chans == 2) {
|
|
put_bits(&s->pb, 1, cpe->common_window);
|
|
if (cpe->common_window) {
|
|
put_ics_info(s, &cpe->ch[0].ics);
|
|
if (s->coder->encode_main_pred)
|
|
s->coder->encode_main_pred(s, &cpe->ch[0]);
|
|
encode_ms_info(&s->pb, cpe);
|
|
if (cpe->ms_mode) ms_mode = 1;
|
|
}
|
|
}
|
|
for (ch = 0; ch < chans; ch++) {
|
|
s->cur_channel = start_ch + ch;
|
|
encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
|
|
}
|
|
start_ch += chans;
|
|
}
|
|
|
|
frame_bits = put_bits_count(&s->pb);
|
|
if (frame_bits <= 6144 * s->channels - 3) {
|
|
s->psy.bitres.bits = frame_bits / s->channels;
|
|
break;
|
|
}
|
|
if (is_mode || ms_mode || tns_mode || pred_mode) {
|
|
for (i = 0; i < s->chan_map[0]; i++) {
|
|
// Must restore coeffs
|
|
chans = tag == TYPE_CPE ? 2 : 1;
|
|
cpe = &s->cpe[i];
|
|
for (ch = 0; ch < chans; ch++)
|
|
memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs));
|
|
}
|
|
}
|
|
|
|
s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
|
|
|
|
} while (1);
|
|
|
|
put_bits(&s->pb, 3, TYPE_END);
|
|
flush_put_bits(&s->pb);
|
|
avctx->frame_bits = put_bits_count(&s->pb);
|
|
|
|
// rate control stuff
|
|
if (!(avctx->flags & AV_CODEC_FLAG_QSCALE)) {
|
|
float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
|
|
s->lambda *= ratio;
|
|
s->lambda = FFMIN(s->lambda, 65536.f);
|
|
}
|
|
|
|
if (!frame)
|
|
s->last_frame++;
|
|
|
|
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
|
|
&avpkt->duration);
|
|
|
|
avpkt->size = put_bits_count(&s->pb) >> 3;
|
|
*got_packet_ptr = 1;
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int aac_encode_end(AVCodecContext *avctx)
|
|
{
|
|
AACEncContext *s = avctx->priv_data;
|
|
|
|
ff_mdct_end(&s->mdct1024);
|
|
ff_mdct_end(&s->mdct128);
|
|
ff_psy_end(&s->psy);
|
|
ff_lpc_end(&s->lpc);
|
|
if (s->psypp)
|
|
ff_psy_preprocess_end(s->psypp);
|
|
av_freep(&s->buffer.samples);
|
|
av_freep(&s->cpe);
|
|
av_freep(&s->fdsp);
|
|
ff_af_queue_close(&s->afq);
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
|
|
{
|
|
int ret = 0;
|
|
|
|
s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
|
|
if (!s->fdsp)
|
|
return AVERROR(ENOMEM);
|
|
|
|
// window init
|
|
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
|
|
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
|
|
ff_init_ff_sine_windows(10);
|
|
ff_init_ff_sine_windows(7);
|
|
|
|
if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0)
|
|
return ret;
|
|
if ((ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) < 0)
|
|
return ret;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
|
|
{
|
|
int ch;
|
|
FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail);
|
|
FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail);
|
|
FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + AV_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
|
|
|
|
for(ch = 0; ch < s->channels; ch++)
|
|
s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
|
|
|
|
return 0;
|
|
alloc_fail:
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
static av_cold int aac_encode_init(AVCodecContext *avctx)
|
|
{
|
|
AACEncContext *s = avctx->priv_data;
|
|
int i, ret = 0;
|
|
const uint8_t *sizes[2];
|
|
uint8_t grouping[AAC_MAX_CHANNELS];
|
|
int lengths[2];
|
|
|
|
avctx->frame_size = 1024;
|
|
|
|
for (i = 0; i < 16; i++)
|
|
if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
|
|
break;
|
|
|
|
s->channels = avctx->channels;
|
|
|
|
ERROR_IF(i == 16 || i >= ff_aac_swb_size_1024_len || i >= ff_aac_swb_size_128_len,
|
|
"Unsupported sample rate %d\n", avctx->sample_rate);
|
|
ERROR_IF(s->channels > AAC_MAX_CHANNELS,
|
|
"Unsupported number of channels: %d\n", s->channels);
|
|
WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
|
|
"Too many bits per frame requested, clamping to max\n");
|
|
if (avctx->profile == FF_PROFILE_AAC_MAIN) {
|
|
s->options.pred = 1;
|
|
} else if ((avctx->profile == FF_PROFILE_AAC_LOW ||
|
|
avctx->profile == FF_PROFILE_UNKNOWN) && s->options.pred) {
|
|
s->profile = 0; /* Main */
|
|
WARN_IF(1, "Prediction requested, changing profile to AAC-Main\n");
|
|
} else if (avctx->profile == FF_PROFILE_AAC_LOW ||
|
|
avctx->profile == FF_PROFILE_UNKNOWN) {
|
|
s->profile = 1; /* Low */
|
|
} else {
|
|
ERROR_IF(1, "Unsupported profile %d\n", avctx->profile);
|
|
}
|
|
|
|
avctx->bit_rate = (int)FFMIN(
|
|
6144 * s->channels / 1024.0 * avctx->sample_rate,
|
|
avctx->bit_rate);
|
|
|
|
s->samplerate_index = i;
|
|
|
|
s->chan_map = aac_chan_configs[s->channels-1];
|
|
|
|
if ((ret = dsp_init(avctx, s)) < 0)
|
|
goto fail;
|
|
|
|
if ((ret = alloc_buffers(avctx, s)) < 0)
|
|
goto fail;
|
|
|
|
avctx->extradata_size = 5;
|
|
put_audio_specific_config(avctx);
|
|
|
|
sizes[0] = ff_aac_swb_size_1024[i];
|
|
sizes[1] = ff_aac_swb_size_128[i];
|
|
lengths[0] = ff_aac_num_swb_1024[i];
|
|
lengths[1] = ff_aac_num_swb_128[i];
|
|
for (i = 0; i < s->chan_map[0]; i++)
|
|
grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
|
|
if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
|
|
s->chan_map[0], grouping)) < 0)
|
|
goto fail;
|
|
s->psypp = ff_psy_preprocess_init(avctx);
|
|
s->coder = &ff_aac_coders[s->options.aac_coder];
|
|
ff_lpc_init(&s->lpc, avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);
|
|
|
|
if (HAVE_MIPSDSPR1)
|
|
ff_aac_coder_init_mips(s);
|
|
|
|
s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
|
|
|
|
ff_aac_tableinit();
|
|
|
|
avctx->initial_padding = 1024;
|
|
ff_af_queue_init(avctx, &s->afq);
|
|
|
|
return 0;
|
|
fail:
|
|
aac_encode_end(avctx);
|
|
return ret;
|
|
}
|
|
|
|
#define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
|
|
static const AVOption aacenc_options[] = {
|
|
{"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
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{"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
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{"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
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{"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
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{"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"},
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{"faac", "FAAC-inspired method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
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{"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
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{"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
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{"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
|
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{"aac_pns", "Perceptual Noise Substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_pns"},
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{"disable", "Disable perceptual noise substitution", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
|
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{"enable", "Enable perceptual noise substitution", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
|
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{"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "intensity_stereo"},
|
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{"disable", "Disable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},
|
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{"enable", "Enable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},
|
|
{"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_tns"},
|
|
{"disable", "Disable temporal noise shaping", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_tns"},
|
|
{"enable", "Enable temporal noise shaping", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_tns"},
|
|
{"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_pred"},
|
|
{"disable", "Disable AAC-Main prediction", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pred"},
|
|
{"enable", "Enable AAC-Main prediction", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pred"},
|
|
{NULL}
|
|
};
|
|
|
|
static const AVClass aacenc_class = {
|
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"AAC encoder",
|
|
av_default_item_name,
|
|
aacenc_options,
|
|
LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
AVCodec ff_aac_encoder = {
|
|
.name = "aac",
|
|
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_AAC,
|
|
.priv_data_size = sizeof(AACEncContext),
|
|
.init = aac_encode_init,
|
|
.encode2 = aac_encode_frame,
|
|
.close = aac_encode_end,
|
|
.supported_samplerates = mpeg4audio_sample_rates,
|
|
.capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY |
|
|
AV_CODEC_CAP_EXPERIMENTAL,
|
|
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
|
|
AV_SAMPLE_FMT_NONE },
|
|
.priv_class = &aacenc_class,
|
|
};
|